[..] |
ASTERISK-08000: [branch] Error in CDRs involving bridges and Local/ channels (call files) |
ASTERISK-08001: Wrong error message in confiugure script when there is a problem with the openh323 installation |
ASTERISK-08002: silence/* files in asterisk-extra-sounds all the same |
ASTERISK-08003: [patch] problem with simple transfer of incoming call |
ASTERISK-08004: [patch] agi option missing from usage warning |
ASTERISK-08005: chan_zap says its ignoring JB settings but it really isn't |
ASTERISK-08006: SVN-oej-jitterbuffer-1.2-r38923 |
ASTERISK-08007: queue.conf option to automatically pause an agent whose status is Busy / In use |
ASTERISK-08008: Segmentation Fault when a call is transfered from a queue |
ASTERISK-08009: Crash in chan_h323 when dialing invalid non existing extension |
ASTERISK-08010: Update to doc/backtrace.txt to reflect 1.4/trunk changes and increase readability |
ASTERISK-08011: chan_sip crashes while trying to find video codecs |
ASTERISK-08012: SIP Segfault with high call setup volume in ast_rtp_lookup_code() |
ASTERISK-08013: SIP paketization |
ASTERISK-08014: asterisk 1.2.12.1 crashes with core several times/week during nightly restart script |
ASTERISK-08015: asterisk crash when SDP contain no description |
ASTERISK-08016: [patch] "list" considered harmful |
ASTERISK-08017: [patch] speeling errors in program comments |
ASTERISK-08018: Asterisk 1.4 99% cpu usage and crashing |
ASTERISK-08019: Segmentation fault on ast_channel_spy_remove |
ASTERISK-08020: [patch] rename app_cdr to app_nocdr, update copyright and doxygen info |
ASTERISK-08021: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) |
ASTERISK-08022: [patch] [1.4] Fix "core show translation" output |
ASTERISK-08023: RFC broken with media stream on INVITE - Report from Ekiga mailing list |
ASTERISK-08024: Disable shell escape '!' (bang) commands in cli |
ASTERISK-08025: [patch] Improve Doxygen output, fix various typos |
ASTERISK-08026: [patch] ast_gethostbyname("192.168.0.1", &hp) fails to set h_addrtype |
ASTERISK-08027: [patch] wav49 format generates 50% compatible files with Windows Media Player |
ASTERISK-08028: dot files in /var/lib/asterisk/moh will cause the dir to be treated as a file - blocking the music on hold to play |
ASTERISK-08029: [patch] Voicemail recording in progress causes asterisk to freak out when file size exceeds 2.1GB |
ASTERISK-08030: macro's appdata can't be longer than 114 chars plus a 3 chars exten |
ASTERISK-08031: Zaptel 1.2.10 will not install on Ubuntu 6.10 |
ASTERISK-08032: [patch] Make redhat init script start asterisk later |
ASTERISK-08033: [patch] IAX Peers output over manager should not be CLI formatted |
ASTERISK-08034: [patch] CONTROL STREAM FILE AGI command is broken |
ASTERISK-08035: [patch] Allow func_curl to substitute channel variables on the passed URL |
ASTERISK-08036: Asterisk crashes when logging with cdr_custom |
ASTERISK-08037: Queue not advancing to next member |
ASTERISK-08038: SIP module deadlocks when PostgreSQL cdr db is overloaded |
ASTERISK-08039: last_message_index is returning the first index |
ASTERISK-08040: voicemail fails when language set to german (de), 1 message present and no german sound files |
ASTERISK-08041: iaxprov.conf in asterisk-1.4.0.beta3 contains tos=lowdelay which is deprecated |
ASTERISK-08042: [patch] Added "loose" option to "joinempty" and "leavewhenempty" |
ASTERISK-08043: BLF not working when call-limit is enabled |
ASTERISK-08044: Client phones nuked with "481 Call leg/transaction does not exist" |
ASTERISK-08045: [branch] chan_iax2.c:3782 iax2_trunk_queue: Maximum trunk data space exceeded |
ASTERISK-08046: issue 0007268 not properly fixed |
ASTERISK-08047: asterisk segfaults at ast_translator_free_path () |
ASTERISK-08048: Dundi Problem with Key generated on hardened system. |
ASTERISK-08049: Signalling levels - TBR21 :- the difference level of high frequency group and low frequency shall be 1 .. |
ASTERISK-08050: memleak in chan_sip |
ASTERISK-08051: After a while of operation, IAX becomes behaving incorrectly, no audio or 1-way, and no-answer |
ASTERISK-08052: g729 codec error loading module Asterisk 1.4 |
ASTERISK-08053: High volume calling results in Asterisk core dump |
ASTERISK-08054: The two files produced by the Monitor application have diferent sizes and lengths, which produces unsynchronized recordings |
ASTERISK-08055: Voicemail copy to multiple boxes fails |
ASTERISK-08056: missing information of the origin exten in the CLI (both with realtime or flat files) |
ASTERISK-08057: Caller ID |
ASTERISK-08058: Replacing a pointless recursion in q921_transmit() |
ASTERISK-08059: [patch] INVITE w/Replaces - Require: header interop issue |
ASTERISK-08060: Asterisk not loading properly on boot |
ASTERISK-08061: [patch] INVITE w/Replaces - Replaces: header incorrectly uri encoded |
ASTERISK-08062: [patch] rtp goes into an error loop on network read error |
ASTERISK-08063: [patch] iax channel goes into an error loop on network read error |
ASTERISK-08064: SLIN codec noise |
ASTERISK-08065: [patch] perl based safe_asterisk |
ASTERISK-08066: [patch] callee->context is always empty |
ASTERISK-08067: iax2 qualify - false "peer unreachable" |
ASTERISK-08068: No such command 'core verbose atleast' |
ASTERISK-08069: SendDTMF() through IAX channel transmits only first digit |
ASTERISK-08070: URIENCODE doesn't handle '#' correctly. Could be others |
ASTERISK-08071: [patch] add checks for calloc calls |
ASTERISK-08072: [Patch] unable to specify "no groups" for groups in config files |
ASTERISK-08073: Failure getting "online status" on softphone |
ASTERISK-08074: [patch] Recording synchronization fails due to bad number of samples correlation in ast_read / ast_write |
ASTERISK-08075: Asterisk crash when PRI fails |
ASTERISK-08076: Cmd ChanIsAvail() does not return predictable status codes. |
ASTERISK-08077: sometimes when user delete all voicemails in their folder, people have trouble getting voicemails |
ASTERISK-08078: break instead of continue in get_sip_pvt_byid_locked() |
ASTERISK-08079: Asterisk system crashs main server |
ASTERISK-08080: app_mixmonitor crashes asterisk |
ASTERISK-08081: improper freing of memory in chan_sip.c::unload_module() ? |
ASTERISK-08082: ooh323c in asterisk-addons might not be redistributable under the GNU GPL License |
ASTERISK-08083: "L" parameter causes asterisk crash |
ASTERISK-08084: [patch] Hebrew (he) support in voicemail |
ASTERISK-08085: SIP sends no hangup |
ASTERISK-08086: PRI de-synchronization |
ASTERISK-08087: Asterisk 'invisible' resending registers after registering successful with a peer |
ASTERISK-08088: Restore autoframing option |
ASTERISK-08089: Frequent Crashes |
ASTERISK-08090: "core show version" fails due to comparison of wrong number of args |
ASTERISK-08091: CallerID Callwaiting appears to be broken |
ASTERISK-08092: [patch] entering a dynamically created conference doesn't observe 'q' (quiet) option |
ASTERISK-08093: [patch] nonce-count value is not added correctry |
ASTERISK-08094: iax.conf and sip.conf bindaddr don't listen on every interface |
ASTERISK-08095: [patch] Refactoring of expression checking implementation |
ASTERISK-08096: core show uptime returns only usage information for the comand |
ASTERISK-08097: voicemail system sending out mangled emails |
ASTERISK-08098: one way audio, when network jitter occur |
ASTERISK-08099: wrong menuselect info |
ASTERISK-08100: wrong menuselect info |
ASTERISK-08101: [patch] A GoSub called from within a Macro clears MACRO_EXTEN, others |
ASTERISK-08102: Quality issues with codec g726 |
ASTERISK-08103: possible memory leak in __sip_ack() |
ASTERISK-08104: ExtenSpy segfault on no given argument to spy from. |
ASTERISK-08105: asterisk and asterisk-sounds conflicts |
ASTERISK-08106: When the phone rings on incoming calls, there are a lot of error messages |
ASTERISK-08107: improper handling of sip_pvt references. |
ASTERISK-08108: Blind transfers crash Asterisk when verbose = 3 |
ASTERISK-08109: [patch] progress_setup and progress_alarm are not properly propagated for outgoing H.323 calls. |
ASTERISK-08110: T.38 Fallback fails |
ASTERISK-08111: call hangup when x=y in L parameter |
ASTERISK-08112: Transfers are not bridging properly |
ASTERISK-08113: ${IAXPEER(targetchannel)} returns null string when called from a SIP (or other non-IAX2) channel |
ASTERISK-08114: Asterisk segfault when trying to include dialplan file with a macro |
ASTERISK-08115: [PATCH] chan_iax2.c remove useless #ifdef |
ASTERISK-08116: still AC_PROG_LD issues... |
ASTERISK-08117: Upon reload of asterisk, cdr_pgsql causes asterisk to seg fault |
ASTERISK-08118: MYSQL will allow table LOCK, but error on a UNLOCK |
ASTERISK-08119: Outgoing SMS: EMS support broken / UDH-bit not set |
ASTERISK-08120: "Dropping voice to exceptionally long" |
ASTERISK-08121: Close Codec Translation Feature |
ASTERISK-08122: Codec in SDP |
ASTERISK-08123: Same that bug 0007351 |
ASTERISK-08124: [patch][post-1.4] new internal API for CLI |
ASTERISK-08125: CLI Command 'module show' is no working |
ASTERISK-08126: Shared Line Appearance sla.conf appending - (dash) character on the context? cause segfault |
ASTERISK-08127: Call transfer or parking failure |
ASTERISK-08128: [patch] H.323 creates channels with nativeformats having MSbit set, which leads to ast_translator_best_choice() chosing it |
ASTERISK-08129: asterisk-1.2.13 Postgres support is enabled when it shouldn't |
ASTERISK-08130: Segfault when inserting a CDR record with primary key overflowing maximum value |
ASTERISK-08131: [patch] Log file rotation on SIGXFSZ doesn't check log file sizes. |
ASTERISK-08132: [patch] change app_amd logging of "AMD using the default parameters" from Notice to Debug |
ASTERISK-08133: [patch] make init file work in SUSE 10 and Redhat too... |
ASTERISK-08134: Reinvite is using local IP of NATed device |
ASTERISK-08135: [patch] Asterisk 1.2 Built in transfer works different from version 1.0 |
ASTERISK-08136: [patch] Warning triggered on "CDR not posted" and "CDR lacks end" when using resetcdr and nocdr apps |
ASTERISK-08137: Timelimit functionality is broken in Dial |
ASTERISK-08138: [patch] Add accuracy range to incoming distinctive ring match |
ASTERISK-08139: Reproduction of bug #6568 in 1.4.0-beta3 |
ASTERISK-08140: Chanspy application in asterisk 1.4 ver crash the asterisk-segmentation fault (core dumped) |
ASTERISK-08141: Asterisk sends CANCEL instead of BYE even if _state is UP |
ASTERISK-08142: [patch] Add OSP support |
ASTERISK-08143: [patch] Upgrade for atxfer behaviour |
ASTERISK-08144: Turning off DTMF Detection or set the sensibility |
ASTERISK-08145: Queue Agent joinempty handling missing an option. |
ASTERISK-08146: [patch] ast_channel_walk_locked or channel_find_locked can "terminate early" |
ASTERISK-08147: Voicemail is leaving open file handles when it create the tmp file |
ASTERISK-08148: On call transfer, need to be able to retrieve SIP Referred-by header from the incoming REFER |
ASTERISK-08149: Passwords are not saved if voicemail users are in an #include file |
ASTERISK-08150: [patch] Remove unused code from manager.c |
ASTERISK-08151: music on hold not random |
ASTERISK-08152: Forwarding "Moved Temporarily" Not Functioning As Expected |
ASTERISK-08153: Potential memory leak in transmit_response_using_temp |
ASTERISK-08154: [patch] voicemail playback via odbc connection gives segfault |
ASTERISK-08155: wrong behavior of 'L(x:y:z)' parameters in Dial application |
ASTERISK-08156: T.38 passthru not invoked when using a Local channel |
ASTERISK-08157: can't call new users |
ASTERISK-08158: It looks like deadlocking of channel |
ASTERISK-08159: powerof(0) can happen with external channel drivers in translate.c |
ASTERISK-08160: IAX2 trunking not enabled in one direction for 'user' rather than 'friend' |
ASTERISK-08161: Bandwidth Requirement |
ASTERISK-08162: inconsistent return checks on handle_request() |
ASTERISK-08163: chan_skinny doesn't send keepalives |
ASTERISK-08164: potential panics induced by app_dial.c::do_forward() |
ASTERISK-08165: "-p: not found" on building |
ASTERISK-08166: sh doesn't like the == operator [PATCH] |
ASTERISK-08167: call/pickupgroups above 32 do not work, even though the docs state otherwise |
ASTERISK-08168: chan_h323 disables fastStart in connect message |
ASTERISK-08169: Wrong ptime cause no audio |
ASTERISK-08170: Removing "Unavailable Message", "Busy Message", "Name" |
ASTERISK-08171: probably useless code in handle_response_register() |
ASTERISK-08172: [patch] /proc/zaptel/1 returns spurious characters at end |
ASTERISK-08173: Hints no longer work in 1.4beta3 |
ASTERISK-08174: ParkedCall does not native bridge. |
ASTERISK-08175: [patch] app_queue device state change race |
ASTERISK-08176: q.931: IntID: Explicit seems not supported by MD110 |
ASTERISK-08177: [patch] Connect Asterisk as a component to a jabber server |
ASTERISK-08178: Limited number of channels |
ASTERISK-08179: Limited number of channels |
ASTERISK-08180: asterisk crashes when transfering zap channel from idefisk to park extension 700 |
ASTERISK-08181: [PATCH] if (debug); printk ... in wcte11xp.c |
ASTERISK-08182: Fix make when make is not GNU make |
ASTERISK-08183: Asterisk process utilizing 100% of CPU time after a P2P'd sip call is hung up |
ASTERISK-08184: Background() application over AGI doesn't return control until file is through playing |
ASTERISK-08185: Called SIP subscriber still ringing even after hanging up the call by calling side |
ASTERISK-08186: Using ODBC voicemail, user intros for unavail, busy and greet are not entered in database |
ASTERISK-08187: IMAP storage in trunk truncates username and password to 3 chars |
ASTERISK-08188: When using IMAP storage, "voicemail show users" does not read message count from imap |
ASTERISK-08189: Potential deadlock in zt_hangup() (with driver interaction...) |
ASTERISK-08190: [patch] OriginateSuccess and OriginateError incomplete |
ASTERISK-08191: sip show inuse does not return anything |
ASTERISK-08192: reference memory after free(). in pbx/pbx_spool.c |
ASTERISK-08193: unable to add new trunks in asteriskNOW |
ASTERISK-08194: bkps directory missing -- tarball creation fails |
ASTERISK-08195: install/runtime support for 586 and lower |
ASTERISK-08196: memory leak , ast_frdup called without free and passthrough codecs used. |
ASTERISK-08197: linux/compiler.h must not be included in 2.6.17 |
ASTERISK-08198: bug with INVAL event |
ASTERISK-08199: Asterisk is sending INVAL packages without a reason |
ASTERISK-08200: [patch] forkcdr does not work as expected |
ASTERISK-08201: 1.4.0b3 crashed during call transfer |
ASTERISK-08202: Transfer on a Polycom phone does not set hint to Idle on transfer completion |
ASTERISK-08203: /dev/zap/pseudo permissions can't be changed |
ASTERISK-08204: Voicemail password problem with users.conf |
ASTERISK-08205: Chanspy whisper does not work as expected |
ASTERISK-08206: Top-level Makefile variable not exported |
ASTERISK-08207: Agents and SIP attended transfers gives warning messages (codec issue) |
ASTERISK-08208: Escaped SIP URI (RFC 3261) doesn't match in dialplan (e.g. "#" with SNOM phones) |
ASTERISK-08209: [patch] Forwarding old voicmail to another user will just be sent to email not to target voicemail |
ASTERISK-08210: G722 audio tarball filename failure for "make" |
ASTERISK-08211: SRV record lookup failing |
ASTERISK-08212: The channel is not hanged up in the right time |
ASTERISK-08213: unexpected ringtone |
ASTERISK-08214: [patch] Allow whisper functionality in Meetme |
ASTERISK-08215: No option in GUI to modify codec prefs in iax.cong and sip.conf |
ASTERISK-08216: [patch] Packet2Packet bridge incompatible with STUN packets processing |
ASTERISK-08217: memory leak at res_features.c |
ASTERISK-08218: Update sip.conf.sample files to show usage of port, and different between bindport |
ASTERISK-08219: [patch] clean up some compile issues on FreeBSD (6.1) |
ASTERISK-08220: Admin password not set for VMplayer version |
ASTERISK-08221: Digium URLs in the admin UI are broken |
ASTERISK-08222: Dialog box for mismatched admin passwords refers to "root" password |
ASTERISK-08223: Localhost login refers to "AsteriskNow" not "AsteriskNOW" |
ASTERISK-08224: Scrollbox on Time Setup screen only one row high; scrollbar unusable |
ASTERISK-08225: AsteriskNOW logo splash screen needed on boot screen |
ASTERISK-08226: [patch] Zaptel trunk fail to compile / install on FreeBSD (6.1) |
ASTERISK-08227: No root password (security problem) |
ASTERISK-08228: Input field validation is timing dependent |
ASTERISK-08229: Input fields consisting of all spaces are allowed |
ASTERISK-08230: Overloaded use of user "password" |
ASTERISK-08231: System allows variable length extensions |
ASTERISK-08232: Sort-by-name option in user list |
ASTERISK-08233: Type-ahead search in user list |
ASTERISK-08234: Tooltip for Conference , "Record conference" is missing |
ASTERISK-08235: Populate call queue extension with next available extension number |
ASTERISK-08236: Make all users agents by default |
ASTERISK-08237: Graceful handling of case where no analog lines are installed |
ASTERISK-08238: On Trunks: "creating new entry" under wrong list |
ASTERISK-08239: Trunks: empty VoIP provider list |
ASTERISK-08240: For SIP providers, help choose closest server |
ASTERISK-08241: System information |
ASTERISK-08242: System information: need GUI version information |
ASTERISK-08243: Backup can create multiple backup files with the same name |
ASTERISK-08244: Backup files can't be downloaded / uploaded |
ASTERISK-08245: Backup: "download configuration backup" |
ASTERISK-08246: Optionstab: typos "atleast" and "donot" |
ASTERISK-08247: [bounty] feature request - QSIG call diversion interop with SIP |
ASTERISK-08248: Support for uploaded files |
ASTERISK-08249: System info tab should provide access to logs of recent error/status messages |
ASTERISK-08250: Clicking on a backup filename reports "404 file not found" |
ASTERISK-08251: No way to restore a backup once created |
ASTERISK-08252: Backup files should summarize the date the backup was created |
ASTERISK-08253: sysinfo displays incorrect information |
ASTERISK-08254: AgentCallbackLogin and SIP hold music doesn't work in user to agent direction. |
ASTERISK-08255: [patch] fixed a typo in the 'dundi-e164-canonical' section |
ASTERISK-08256: Custom-voip provider has provider = iaxtel |
ASTERISK-08257: [patch] compiling and installing asterisk-gui on FreeBSD (6.1) |
ASTERISK-08258: No DTMF tone with chan_misdn |
ASTERISK-08259: [patch] asterisk-addons doesn't respect --prefix |
ASTERISK-08260: having a video play for auto attendant for those with video phones |
ASTERISK-08261: unable to log in to gui using konqueror web browser |
ASTERISK-08262: [patch] Bringing back to Makefile pridump, pritest and testprilib binaries |
ASTERISK-08263: panels do not respond to mouse click in konqueror |
ASTERISK-08264: [patch] snmp.txt says it needs more libraries, but doesn't tell us which ones |
ASTERISK-08265: [patch] odbcstorage.txt references command in voicemail.conf that is not in .sample file |
ASTERISK-08266: asterisk-gui setup.html crash asterisk with included file on extensions.conf from nfs |
ASTERISK-08267: chan_gtalk and chan_jingle outgoing calls remain locked without calling anything |
ASTERISK-08268: asterisk-ooh323c from asterisk-addons svn trunk don't compile with latest asterisk svn trunk |
ASTERISK-08269: Crash when having more than 10 IAX registrations per second |
ASTERISK-08270: [patch] ES-EN translation of code comments in fskmodem.c |
ASTERISK-08271: On high call volume, Asterisk starts reporting: cause 34 - Circuit/channel congestion |
ASTERISK-08272: [patch] fix check for curl-config and removed reduntant code |
ASTERISK-08273: Webserver is saying "Invalid/Unknown Command" instead of saying "already Logged In" |
ASTERISK-08274: Use of voicemail ODBC storage not possible with Postgresql |
ASTERISK-08275: Possible issue with Early Media still in invitestate-1.4 branch (and thus other Asterisk branches) |
ASTERISK-08276: [patch] Voicemail does not playback via ODBC voicemail storage with Postgresql database |
ASTERISK-08277: /etc/init.d/asterisk and/or safe_asterisk breaks cron/anacron (Debian) |
ASTERISK-08278: [patch] voicemail with volgain leaves behind temp file |
ASTERISK-08279: [patch] add option to disable announcement of queue position |
ASTERISK-08280: [patch] aditional manager commands DBDel and DBDelTree |
ASTERISK-08281: Enhance CUT function to allow range variable |
ASTERISK-08282: Goto fails in applicationmap (res_features) |
ASTERISK-08283: Asterisk crashes when updating state on a expired realtime peer (res_config_mysql) |
ASTERISK-08284: app_dial setting callerid to ${EXTEN} too soon |
ASTERISK-08285: Reading DTMF fails on IAX2 |
ASTERISK-08286: asterisk-gui v150 fails to "make" under OpenSUSE 10.1 64-bit |
ASTERISK-08287: extensions limited to 4 digits in "updated" AsteriskNOW |
ASTERISK-08288: [patch] Via: header may contain multiple values |
ASTERISK-08289: Permission denied, opening /dev/snd/controlC0, even if the user, under which asterisk runs as, has access to it. |
ASTERISK-08290: [patch] Incorrect variable name 'rtignoreexpire' in iax.conf.sample |
ASTERISK-08291: multiples refers problem |
ASTERISK-08292: 487 retransmits are not ACKed |
ASTERISK-08293: Inbound Caller Does Not Hear Ringing |
ASTERISK-08294: [patch] Say Digits does not work correctly |
ASTERISK-08295: STRFTIME() requires an argument, but function description does not reflect that |
ASTERISK-08296: [patch] extend IAX2 to support OSP protocol |
ASTERISK-08297: Manager redirect hangs up on calls in AGI |
ASTERISK-08298: Asterisk server crashes on 'undefined symbol: ast_adsi_available' in VoiceMailMain if res_adsi.so is not loaded |
ASTERISK-08299: Need FreeWorld Dialup, Broadvoice, Sipura install procedures for AsteriskNOW |
ASTERISK-08300: T.38 negotiation fails when h263 is enabled |
ASTERISK-08301: [patch] Caller ID not set in CDR |
ASTERISK-08302: zaptel module (wcte11xp) makes asterisk unable to reproduce audio messages |
ASTERISK-08303: Hard phone issue: |
ASTERISK-08304: [patch] segmentation fault when unable to play files |
ASTERISK-08305: [patch] When using L option on Dial, instead of warning asterisk disconnects the call |
ASTERISK-08306: Incoming RDNIS redirecting number variable left unset on EuroISDN |
ASTERISK-08307: [patch] OpenBSD 4.0 "make" fails |
ASTERISK-08308: Hard phone issue: |
ASTERISK-08309: Asterisk Ignoring port parameter in sip.conf |
ASTERISK-08310: [patch] DTMF fails and one-way-audio after negative timestamp. |
ASTERISK-08311: Asterisk dumps core when briding in p2p mode |
ASTERISK-08312: Playback and StopPlayback Manager commands |
ASTERISK-08313: Queue does not work with SIP gateway |
ASTERISK-08314: Need IE, Safari support |
ASTERISK-08315: [patch] ast_app_getdata() without any prompts to play |
ASTERISK-08316: manager: split / redirect call |
ASTERISK-08317: show iax peer details in manager and cli |
ASTERISK-08318: [patch] introduce distinction between overlap-dial in sending/receiving mode |
ASTERISK-08319: even though configure is ran with --prefix, make install tries to mkdir /var/lib/asterisk |
ASTERISK-08320: Large SIP messages are truncated to 4096 bytes |
ASTERISK-08321: 'admin' console login password not set in vmplayer version |
ASTERISK-08322: "Local extension settings" should be on the Admin options panel |
ASTERISK-08323: Download backup should be streamlined |
ASTERISK-08324: "next available" extension numbers should conform to extension length |
ASTERISK-08325: Misspelling in Service providers tab |
ASTERISK-08326: Asterisk stays in the audio path if "t" option in Dial is used |
ASTERISK-08327: Action originate with app Set don't call |
ASTERISK-08328: Asterisk crashes if it can not find file that it just recorded |
ASTERISK-08329: Authentication using contact user from registration instead of specified |
ASTERISK-08330: Adding Rhino Card Installation to Core Installer |
ASTERISK-08331: Email notification has blank CIDNAME (should use CIDNUM or an "unknown caller" if empty but doesn't). |
ASTERISK-08332: [patch] allow asterisk database to be used for static configuration for files |
ASTERISK-08333: [patch] implement basic "Shared Lines" functionality |
ASTERISK-08334: Chanell always showsa as Zap/0-0 regardless of actual channel in use. |
ASTERISK-08335: REFER not working with Cisco hardware when doing local attended call transfer |
ASTERISK-08336: SIP HOLD propegation to AST_CONTROL |
ASTERISK-08337: Asterisk core when busy in a zap call |
ASTERISK-08338: Call processing stops during reload on systems with large dialplan |
ASTERISK-08339: [patch] MWI Error with NOTIFY on Cisco IP phone firmware > 8.0.3 |
ASTERISK-08340: Spandsp + Ast 1.4B3 - crash on incoming RXFAX |
ASTERISK-08341: Unable to join queue |
ASTERISK-08342: Asterisk randomly crashes |
ASTERISK-08343: Reg asterisk-1.4 installation problem-undefined rerference to'ast_copy+_string' |
ASTERISK-08344: [patch] Limit on simultaneous calls for queue members |
ASTERISK-08345: building 1.4.0-beta4 fails when linking chan_zap (missing libpri dependency alert) |
ASTERISK-08346: [patch] Update to use the new ast_channel_alloc format |
ASTERISK-08347: Local channels hanging |
ASTERISK-08348: Call waiting Notification from PRI |
ASTERISK-08349: [patch] extend app_SMS to support protocol 2 (in use in Italy, Spain, xxx) |
ASTERISK-08350: [patch] Caller Id and Message Waiting Indicator problems |
ASTERISK-08351: Display error in Agents list |
ASTERISK-08352: Goto Exten fails when used as a step in Voice Menus |
ASTERISK-08353: Festival Application Hangs Call |
ASTERISK-08354: asterisk drops call long call |
ASTERISK-08355: chan_sip does not handle 504 "Service Unavailable" case |
ASTERISK-08356: [patch] dstchannel in cdr is empty when transfer call |
ASTERISK-08357: [patch] make field names configurable in cdr_addon_mysql |
ASTERISK-08358: [patch] Wrong Coding of Name in 'To:' header of Emails |
ASTERISK-08359: [patch] Asterisk doesn't send CANCEL before Ringing |
ASTERISK-08360: SIP, dtmf-relay, feature key presses being ignored |
ASTERISK-08361: RetryDial does not properly support the G() Dial option |
ASTERISK-08362: SIP bug in handling invitestate |
ASTERISK-08363: [branch] improper computation of Content-Length in add_t38_sdp() |
ASTERISK-08364: [patch] logic of handle_common_options() in channel_sip.c (2 issues) |
ASTERISK-08365: [patch] C++ modules fails to compile, strings.h is not C++ clean |
ASTERISK-08366: Meetme conference application randomly crashing with app_meetme.c errors |
ASTERISK-08367: [patch] threads syncronization |
ASTERISK-08368: ./configure --prefix... ignored. |
ASTERISK-08369: No OriginateSuccess or OriginateFailure event after a Originate command. |
ASTERISK-08370: Add link in GUI directly to bug tracking system to report problems |
ASTERISK-08371: Support 12-hour format clock (in addition to 24-hour/military clock currently supported) |
ASTERISK-08372: Allow for GUI updating from within the GUI |
ASTERISK-08373: Sound for "October" is said as "Tober" |
ASTERISK-08374: [patch] likely memory leak in app_dial (trunk, 1.4 and 1.2) |
ASTERISK-08375: [patch] Added RTCP Manager Events to rtp.c |
ASTERISK-08376: Asterisk 1.4.0-beta4 compile errors on Fedora Core 6 |
ASTERISK-08377: "System Info" tab reports "file not found" dialog box |
ASTERISK-08378: System Info / Logs report "404 Not Found" |
ASTERISK-08379: Typo on screen 1 of 7 of setup wizard "Verify Analog ports" |
ASTERISK-08380: "starting point" of allocation should be consistent with extension length selection |
ASTERISK-08381: Setup wizard: Calling rules step (5 of 7); should default to provider |
ASTERISK-08382: Setup calling rules (5 of 7): Can't "save" edits to calling rules |
ASTERISK-08383: Tab style in "Options" and "System Info" should match |
ASTERISK-08384: Calling Rules -- dialog box cascade for "undefined" items |
ASTERISK-08385: "F2 Incorporated"'s logo is broken |
ASTERISK-08386: [patch] make message length configurable per user instead of only globally |
ASTERISK-08387: [patch] Allow voicemail to use an external app and smdi at the same time. |
ASTERISK-08388: mis-spelling in doc/snmp.txt |
ASTERISK-08389: build fails on snmp/agent.c |
ASTERISK-08390: RFC 2833 dialtone packets out of order can cause extra digits to be reported |
ASTERISK-08391: Dialog box for missing password on login has wrong text |
ASTERISK-08392: Several problems with setup wizard login screen |
ASTERISK-08393: Possible to drive UI to state strange state |
ASTERISK-08394: Copyrights need updating |
ASTERISK-08395: Updated text needed for "Service Providers" tool tip |
ASTERISK-08396: Incoming calls tab is blank |
ASTERISK-08397: Admin password screen still refers to "Business Edition"T |
ASTERISK-08398: System failed to detect analog ports |
ASTERISK-08399: asterisk fails to cross compile for arm |
ASTERISK-08400: Bug 0006181 still exists. |
ASTERISK-08401: iax2 crash on transfer |
ASTERISK-08402: Trademark link inside the footer is broken |
ASTERISK-08403: setup wizard stuck on step 2 |
ASTERISK-08404: Legal Information link broken |
ASTERISK-08405: Problem receiving calls from BroadWorks |
ASTERISK-08406: IAX2 outgoing calls not working |
ASTERISK-08407: Music on Hold for Call Queues |
ASTERISK-08408: Asterisk segfaults (core dumped) at startup because of corrupted astdb |
ASTERISK-08409: The UI should report to the user attempted use of incompatible browsers |
ASTERISK-08410: Active channels not cleaned upon entering an UNREACHABLE state |
ASTERISK-08411: No inbound CallerID when Distinctive Ring Detection is enabled. |
ASTERISK-08412: No dialtone on analog FXS ports |
ASTERISK-08413: Dead AGI : not able to use StartMusicOnHold application |
ASTERISK-08414: Can't make rpm in Asterisk 1.2.14 due to missing files in asterisk.spec |
ASTERISK-08415: Remove silence files from Asterisk Extra Sounds, since Core sounds have them since 1.4.b4 |
ASTERISK-08416: Pickup using g729 |
ASTERISK-08417: [patch] canreinvite = nonat does not cause packet2packet bridge |
ASTERISK-08418: [patch] libiax2 wrong timestamp |
ASTERISK-08419: Parking causing crashes |
ASTERISK-08420: segfault at irregular interval |
ASTERISK-08421: [patch] Add a jabber text receiver application, and make Asterisk a Gtalk to phone gateway |
ASTERISK-08422: Possible memory leak doing only inbound SIP handling |
ASTERISK-08423: Cut function requires | delimiter |
ASTERISK-08424: Parked Calls drop immediately |
ASTERISK-08425: [patch] enable MPEG4 Part 2 video codec pass-through |
ASTERISK-08426: [patch] option to restrict manager users to a single simultaneous login |
ASTERISK-08427: func_math.c: iaction set to GTFUNCTION when the first char of operator is '=' |
ASTERISK-08428: TDM400+SIP paketizations |
ASTERISK-08429: Stringfield pool corruption, segmentation fault during free. |
ASTERISK-08430: Blind Transfer does not fail when destination is unreachable |
ASTERISK-08431: Asterisk auto-creates meetme conference when not ask to do so. |
ASTERISK-08432: [patch] libpri Makefile doesn't use full path to restorecon |
ASTERISK-08433: [patch] zaptel Makefile doesn't use full path to restorecon |
ASTERISK-08434: MOH doesn't resume from where it was left off |
ASTERISK-08435: [patch] app_page.so 's' option flag - skip adding channel to meetme if devicestatus != not in use |
ASTERISK-08436: vm-youhaveno sound is missing in spanish |
ASTERISK-08437: Sound 1M is not found in Spanish even though it exists |
ASTERISK-08438: [patch] Shell Dialplan Function, returns output |
ASTERISK-08439: Asterisk does not reinvite peer for G.711 after T.38 negotiated failed with a "488" Event |
ASTERISK-08440: GetConfig + #include causes segfault |
ASTERISK-08441: [patch] coding guidelines compliance for main/*.c |
ASTERISK-08442: billsec is 0 even when the call is answered |
ASTERISK-08443: Asterisk Manager Interface Reload segfaults if clients are connected. |
ASTERISK-08444: 1.4.0 release UPGRADE.txt lists old show channels concise method whcich doesn't work |
ASTERISK-08445: Blind Transfer within DIAL in DeadAGI does not work |
ASTERISK-08446: [patch] Fix bad handling of #include directives from manager GetConfig/UpdateConfig |
ASTERISK-08447: * eats all CPU. |
ASTERISK-08448: Crash on blind transfer of an incoming call (queue) |
ASTERISK-08449: Asterisk 1.2.7.1 Crashing |
ASTERISK-08450: Check type selection for sizes |
ASTERISK-08451: Frequent seg fault in ast_cdr_alloc() at cdr.c:438 |
ASTERISK-08452: verbose info with no if (option_debug) so it always shows |
ASTERISK-08453: /etc/sudoers file becomes corrupt |
ASTERISK-08454: Zap lines are always named as Zap/0-0 |
ASTERISK-08455: Lithuanian syntax for ast_say_number_full |
ASTERISK-08456: [patch] SQLite3 CDR Backend |
ASTERISK-08457: IAX2 appears to deadlock on OS X |
ASTERISK-08458: Cannot pass audio after transfer, intermittent segfault |
ASTERISK-08459: pbx_load_users adds IAX instead of IAX2 to dial string and hint |
ASTERISK-08460: Segfault on 1.4.0 with E1s |
ASTERISK-08461: [patch] openh323 of Debian not detected by autoconf |
ASTERISK-08462: SIP User-Agent string should display Asterisk version |
ASTERISK-08463: IAX2 configuration parser reverses general and specific parameters when loading users |
ASTERISK-08464: Sounds played in MEETME_EXIT_CONTEXT of conference stutters. |
ASTERISK-08465: app_wait rounds down delay to nearest whole number |
ASTERISK-08466: queue causes a crash |
ASTERISK-08467: Call dropped with "FRAME_CONTROL (5)" message |
ASTERISK-08468: Crash 20 seconds after reload app_queue.so |
ASTERISK-08469: Replace not working properly? |
ASTERISK-08470: 302 Redirect not working? |
ASTERISK-08471: Extensions. Only digits allowed in GUI, other characters ok if file users.conf. |
ASTERISK-08472: random clicking of pages in GUI produces a crash in manager |
ASTERISK-08473: Asterisk GUI does not output Asterisk Log |
ASTERISK-08474: AsteriskGUI tells wrong version in system info tab |
ASTERISK-08475: Select MoH thru GUI |
ASTERISK-08476: Calling Rules Limited to Dialpan1 |
ASTERISK-08477: Transfering of calls does not work in 1.4 through chan_agent |
ASTERISK-08478: MusicOnHold application drops call after exactly one minute |
ASTERISK-08479: mixminotor don't record sounds played to callee |
ASTERISK-08480: Seg fault on call made after manager connection |
ASTERISK-08481: Asterisk 1.4 crash for some reason (iax stuff) |
ASTERISK-08482: Attend transfer with internal Polycom tranfer method does not show original caller id |
ASTERISK-08483: Origdate field in voicemail msg0000.txt has an extra space |
ASTERISK-08484: Asterisk crashes in bridge_p2p_rtp_write |
ASTERISK-08485: [patch] calltime duration with L() keeps warning for all last seconds |
ASTERISK-08486: file include directive doesn't work in extensions.conf |
ASTERISK-08487: Dynamically Generate list of sound files |
ASTERISK-08488: codec_zap no longer compiles as it can't find zaptel/zaptel.h in nonstandard place |
ASTERISK-08489: Users can't make IAX2 calls |
ASTERISK-08490: wcte11xp and wcte4xxp and wctdm inside in one box work issue |
ASTERISK-08491: Stack gets confused when Protocol error received after SETUP |
ASTERISK-08492: Question about the Issue 5853 |
ASTERISK-08493: Can't compile with debian unstable due to genksyms not found |
ASTERISK-08494: [patch] RELEASE COMPLETE is sent in state ACTIVE - leaves the trunk busy |
ASTERISK-08495: Assign extensions to queues |
ASTERISK-08496: [patch] transmit_state_notify for DIALOG_INFO_XML wrong |
ASTERISK-08497: asterisk reinvites to G.711 after a T.38 negotiation - fax fails depending on ATA config |
ASTERISK-08498: Allow a reason to be specified when pausing an agent. |
ASTERISK-08499: [patch] Allow a reason to be specified when pausing an agent. |
ASTERISK-08500: [patch] zapata.conf cannot reset usedistinctiveringdetection and distinctiveringaftercid |
ASTERISK-08501: [patch] hasmanager in users.conf has no effect |
ASTERISK-08502: [branch] PlayDTMF with a non-existing channel will cause segmentation fault |
ASTERISK-08503: Console interface (-r) stops operating, with logging |
ASTERISK-08504: When using the asterisk manager to originate a call the billsec field in CDR's is set to 0 |
ASTERISK-08505: 1.4.0 has teardown/hangup issues after attended transfer |
ASTERISK-08506: [patch] Fix app_read to play multiple files |
ASTERISK-08507: Registration acknowledgement incorrectly handles missing refresh value |
ASTERISK-08508: Asterisk 1.4.0 Hint status not updated afer HOLD state |
ASTERISK-08509: callerid.c loses name when returning PRIVATE_NUMBER flag |
ASTERISK-08510: Inactivity TimeOut => panel opens blank frame |
ASTERISK-08511: Sip phones don't appear in users list |
ASTERISK-08512: Attended Transfer with Polycom handset results in stuck call. |
ASTERISK-08513: Compile of SVN zaptel failed |
ASTERISK-08514: Codec_zap SVN failed to build |
ASTERISK-08515: ALSA output causes Seg Fault |
ASTERISK-08516: ztcodec not in menuselect selection |
ASTERISK-08517: Asterisk crashed with core dump in reqprep |
ASTERISK-08518: Spying by an IAX2 endpoint kills SIP calls |
ASTERISK-08519: password changes in voicemail does not work (using realtime) |
ASTERISK-08520: "Zaptel transcoder support loaded" message messes up screen and obscures button |
ASTERISK-08521: Wav49 support appears broken |
ASTERISK-08522: [patch] Voicemail fails to authenticate users created from users.conf |
ASTERISK-08523: libnsl no necesary in BSD's |
ASTERISK-08524: ZAP channels dies after a while (a week or close) of asterisk usage |
ASTERISK-08525: setMusiconHold option does not show MOH files. |
ASTERISK-08526: Asterisk crashes in call centre environment several times a week |
ASTERISK-08527: When configured with hasiax = yes IAX is dialed instead of IAX2 |
ASTERISK-08528: asterisk will not handoff RTP!!! |
ASTERISK-08529: SIP message 420 Bad extension sent out malformed. |
ASTERISK-08530: MeetMeJoin manager event sent when leaving room |
ASTERISK-08531: Extension length problems with voice mail |
ASTERISK-08532: System happy to assign multiple extensions to the same analog line |
ASTERISK-08533: "Called id" text in user extensions setup wizard should be "caller id" |
ASTERISK-08534: Asterisk fails to build from 1.4 branch on Fedora Core 5 due to a missing gsm.h |
ASTERISK-08535: Attended transfers to parking broken |
ASTERISK-08536: asterisk config options for external libs (ssl, qt3, ncurses, ...) are not followed |
ASTERISK-08537: Latest SVN crashes asterisk with SIP registion and res_config_pgsql.(pgsql) |
ASTERISK-08538: Module cdr_manager.so doesn't load if cdr_manager.conf exists. |
ASTERISK-08539: brazilian portuguese syntax in voicemail with problems (pt_BR) |
ASTERISK-08540: Crash with queues and transfers on a call center eviroment. |
ASTERISK-08541: Cannot create incoming call route in AsteriskNOW - GUI does not store it. |
ASTERISK-08542: G729 No path to translate. |
ASTERISK-08543: G729 No path to translate. |
ASTERISK-08544: Code error in chan_sip.c |
ASTERISK-08545: Calls coming on off of a AgentCallbackLogin() queue get dropped upon transfer |
ASTERISK-08546: GUI keeps kicking users out after getting to wizard |
ASTERISK-08547: [patch] Properly handle tempates on config read/write |
ASTERISK-08548: Audio files incorrect for SayNumber with 13..14..15.. |
ASTERISK-08549: Attended transfers using Polycom phones create 100% CPU utilization |
ASTERISK-08550: [patch] SQLite3 resource |
ASTERISK-08551: Segmentation fault (core dumped) when try to use the g729 codec |
ASTERISK-08552: PIN Code and Administrator PIN Code Conferencing are ignored for app_meetme |
ASTERISK-08553: SVN rev 1806 fail on compile |
ASTERISK-08554: SIP message 420 Bad extension sent out malformed. |
ASTERISK-08555: Channel h323 has error in code (invalid function using -> ast_append_ha) |
ASTERISK-08556: Incorrect parsing of IAX2 video frames |
ASTERISK-08557: [patch] performing a goto / gotoif / gotoiftime in the h extension changes the dst field of the cdr |
ASTERISK-08558: chan_sip doesn't recognize localnet |
ASTERISK-08559: [patch] While dtmfmode set to inband asterisk still negotiates rfc2833 mode, but fail to recognize digits |
ASTERISK-08560: hints on SIP-accounts don't work |
ASTERISK-08561: Check PEER first |
ASTERISK-08562: Iaxy wont give dial tone |
ASTERISK-08563: [patch] ast_load_realtime calls ast_load_realtime_all with wrong parameters |
ASTERISK-08564: DTMF no longer recongnized on ParkedCalls |
ASTERISK-08565: Manager Interface no longer recognizes the Login action |
ASTERISK-08566: Last few daily builds of 1.4-svn have produced distorted audio in SIP |
ASTERISK-08567: DTMF outpulse bug |
ASTERISK-08568: [patch] two minor bugfixes for voicemail (one for IMAP storage, one for email notifications) |
ASTERISK-08569: IMAP storage does not work with c-client 2006 |
ASTERISK-08570: unload res_snmp will cause a segfault |
ASTERISK-08571: Chan H323 will not load |
ASTERISK-08572: Channel ooh323 will not appear in channeltypes |
ASTERISK-08573: [patch] Store vm password in external file |
ASTERISK-08574: including a shipped header file based on includepath search doesn't make sense |
ASTERISK-08575: dtmf digits are not transmited reliably from h323 |
ASTERISK-08576: Outbound DTMF Fails with Sipgate using RFC2833 |
ASTERISK-08577: No more registry after a timeout |
ASTERISK-08578: Dial() sending to Macro upon connect |
ASTERISK-08579: [patch] imap storage does not work in conjunction with realtime voicemail |
ASTERISK-08580: During the install process, "make samples" assumes that you have installed the gsm sound files |
ASTERISK-08581: Asterisk doesn't use externip if private IP is outside localnet |
ASTERISK-08582: [patch] GetGroupCount causes a Seg Fault! |
ASTERISK-08583: GUI Voicemenu does not retain keypress 'GoToMenu' events (Can't get it to hold on to a menu tree.) |
ASTERISK-08584: [patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation |
ASTERISK-08585: Queue commands 'i' option |
ASTERISK-08586: Wizard wiped out call rules on first setup |
ASTERISK-08587: [patch] allow the * as the exit dtmf |
ASTERISK-08588: [patch] Know on which channel a stream occurs |
ASTERISK-08589: Add g729 pass-thru support for Sigma Designs boards support |
ASTERISK-08590: accountcode variable is not passed forward on a transfer |
ASTERISK-08591: RFC2833 DTMF "Zero" missing inner RTP packets |
ASTERISK-08592: Additional checks for option_debug |
ASTERISK-08593: r50032 broke DTMF (rfc2833) |
ASTERISK-08594: 500 ms delay on answer introduced channel locking issue |
ASTERISK-08595: Race condition in app_meetme when using the 'e' (empty) and 'd' (dynamic) option and two calls arrive at the "same" time. |
ASTERISK-08596: ael2 reload nukes subscription tables |
ASTERISK-08597: Addition of Transfer and Flash-Hook-Transfer functions into Gui |
ASTERISK-08598: Voice Menu Config does not retain extensions |
ASTERISK-08599: mohsuggest doesn't work |
ASTERISK-08600: ael2 does not support extension in macro |
ASTERISK-08601: Seg.fault when parking a call via an extenension |
ASTERISK-08602: not able to send/receive calls after upgrade |
ASTERISK-08603: [patch] In Realtime Queues, dynamic queue members do not always load the members correctly |
ASTERISK-08604: d-channel is hardcoded to 24 and 16 for PRI T1 and E1 respectively |
ASTERISK-08605: tab isnt working for module load |
ASTERISK-08606: [patch] core show channeltype foo |
ASTERISK-08607: Using Mixmonitor app eat's a lot of memory on > = 10 Simultaneous call |
ASTERISK-08608: [branch] First patch for the CLI filtering |
ASTERISK-08609: Core dumped when too many calls are queued |
ASTERISK-08610: memory leak on command line completion |
ASTERISK-08611: manager authentication does not work |
ASTERISK-08612: many service provider - missing scrollbar |
ASTERISK-08613: contact information should be in users.conf - no incoming calls |
ASTERISK-08614: nat=no is not RFC 3261 compliant regarding sending responses |
ASTERISK-08615: Remote-Party-ID |
ASTERISK-08616: [branch] Don't rotate ODBC log |
ASTERISK-08617: Voicemail recordings not getting saved |
ASTERISK-08618: [patch] Add option to revert old ChanIsAvail() with 's' option behavior |
ASTERISK-08619: GUI - "Buy Now" button non-functional |
ASTERISK-08620: GUI - "Incoming Call Rule" Notice stays after configuration |
ASTERISK-08621: GUI - "Calling Rules" Pattern match adds an extra digit |
ASTERISK-08622: GUI - "Calling Rules" save option does not appear if you only edit the "follow by [ ] digits" box |
ASTERISK-08623: [patch] new channel driver : chan_unistim for Nortel Unistim IP phones |
ASTERISK-08624: Dell 2950 - AsteriskNow will not install using gui |
ASTERISK-08625: Voicemail w/ odbc fails under certain circumstances |
ASTERISK-08626: List of prompts with cut off audio |
ASTERISK-08627: Installer Static IP check 'fails' valid IP. |
ASTERISK-08628: iwconfig missing from install. |
ASTERISK-08629: Wrong 'TYPE' used for wireless card in ifcfg-ethx |
ASTERISK-08630: Voicemail with IMAP storage sends two email messages |
ASTERISK-08631: "BuyNow" button is not fixed on the screen |
ASTERISK-08632: "BuyNow" button does not appear to link to a working Web page |
ASTERISK-08633: Service provider password field shows password as clear text |
ASTERISK-08634: Outbound calling doesn't work on GUI / Asterisk NOW |
ASTERISK-08635: System info / log doesn't filter out extraneous characters that may affect HTML formatting |
ASTERISK-08636: Incoming call rules table not correctly formatted immediately after adding a rule |
ASTERISK-08637: Incoming call rules table not correctly formatted immediately after adding a rule |
ASTERISK-08638: Incoming call rules table not correctly formatted immediately after adding a rule |
ASTERISK-08639: Incoming call rule to route to an extension does not appear to work |
ASTERISK-08640: Accessing HTML Manager Event Interface causes Segmentation Fault |
ASTERISK-08641: TYPO: Makefile looks for .bz instead of .gz |
ASTERISK-08642: Accessing HTML Manager Event Interface causes Segmentation Fault |
ASTERISK-08643: Recording mixing problem with asterisk |
ASTERISK-08644: Calling Rules no longer display since deleting 1 rule - Asterisk Now GUI |
ASTERISK-08645: Asterisk sometimes hangs up channel on Manager Redirect command |
ASTERISK-08646: Asterisk crashes randomly when a transfers from queue member to an SIP extension with spies attached. |
ASTERISK-08647: Zaptel tone indication for congestion wrong for tonezone .au |
ASTERISK-08648: Segfault at CLI |
ASTERISK-08649: [patch] Variable used after it was free() |
ASTERISK-08650: * segfault on module unload chan_sip.so |
ASTERISK-08651: stray DTMF (rfc2833) during conversations |
ASTERISK-08652: doens't start moh when on hold |
ASTERISK-08653: [patch] remove 2 identical checks. |
ASTERISK-08654: No IP given in Console Menu if eth0 is not enabled. |
ASTERISK-08655: SVN trunk is out of date |
ASTERISK-08656: SVN out of date |
ASTERISK-08657: Paging causes sip phone to ring instead of auto answering |
ASTERISK-08658: No audio when peer sends IPV4 0.0.0.0 in sdp |
ASTERISK-08659: inuse counter not decreased after hanging up a call during which caller is put on hold |
ASTERISK-08660: echo while dialing through zaptel card |
ASTERISK-08661: Dial's G option causes strange behavior in some circustances. |
ASTERISK-08662: zap channel hangup is not detected if it occurs before the line is answered |
ASTERISK-08663: Bus error when re-reading config files |
ASTERISK-08664: make vim syntax highlight config files in the /etc/asterisk directory |
ASTERISK-08665: [patch][moremanager branch] IAX had no manager command to list peers in proper format |
ASTERISK-08666: Random segfault when two channels are simultaneously Redirect'ed into a conference |
ASTERISK-08667: options from config template take precedence over per user options (iax.conf) |
ASTERISK-08668: [patch] chan_cellphone - use mobile phones with Bluetooth as FXO devices in * |
ASTERISK-08669: Dumps core at usecount, module format_mp3 |
ASTERISK-08670: EAGI buffer overflow in IPC corrupts sound transfer |
ASTERISK-08671: [patch] deadlock occurs when spy leaves session using ChanSpy due to inverse locking order |
ASTERISK-08672: SIP conversations drop after 30 seconds, started about 72 hours ago with no changes in network. Strangely similar to 0008193 |
ASTERISK-08673: [patch] IMAP Voicemail reports mailbox full if IMAP server does not report quota |
ASTERISK-08674: [post-1.4][branch] CLI command audit |
ASTERISK-08675: Privacy screening mode doesn't answer call before recording. Should do it. |
ASTERISK-08676: Call Forward closing the second cdr |
ASTERISK-08677: [patch] libedit configure for crosscompilation |
ASTERISK-08678: when getting incoming call from zap you cannt put on hold or park the call |
ASTERISK-08679: codec_zap.c failed to compile on asterisk 1.2 |
ASTERISK-08680: cant start * on PowerPC |
ASTERISK-08681: New Features |
ASTERISK-08682: GUI does not work properly in Opera. |
ASTERISK-08683: GUI not working at all |
ASTERISK-08684: Passing a channel variable to func_odbc with comma in double quotes is parsed as two values |
ASTERISK-08685: DTMF modes are bot getting translated in P2P bridging mode |
ASTERISK-08686: Duplication of commands created by changes in revision 47051 |
ASTERISK-08687: Zero length string papameters while logging in build_user function |
ASTERISK-08688: pick up feature in trixbox |
ASTERISK-08689: codec_zap fails to compile on Fedora Core 6 |
ASTERISK-08690: Incorrect SDP in header when not native bridging |
ASTERISK-08691: Documentation Update for pgsql table setup |
ASTERISK-08692: inUse counter not decremented after hanging up a call which is on hold |
ASTERISK-08693: DTMF translation from rfc2833 to inband is not reliable (no native bridging) |
ASTERISK-08694: module unload app_playback.so twice will segfault * |
ASTERISK-08695: After ForkCDR AGI cant set CDR(userfield) |
ASTERISK-08696: transfer is decline |
ASTERISK-08697: [patch] paused status missing from realtime queue members |
ASTERISK-08698: Revision 53033 displays SVN-trunk-r53002 |
ASTERISK-08699: gui install.html no menu , no active button |
ASTERISK-08700: Asterisk accepts RTP from random endpoints |
ASTERISK-08701: Pressing [Alt][F9] before the final menu pops up causes unperdicable results |
ASTERISK-08702: RTP debug prints negative values |
ASTERISK-08703: [patch] app_userevent sends an extra blank line (\r\n\r\n) after events |
ASTERISK-08704: Selecting shutdown on AsteriskNOW Live Beta 1.4 doesn't complete |
ASTERISK-08705: [patch] Channel/Thread deadlock with heavy bi-directional traffic on PRI - GLARE! |
ASTERISK-08706: Asterisk doesn't see a bridged call even though the call goes through |
ASTERISK-08707: glob as a file name in error message |
ASTERISK-08708: [patch] asterisk -F option missing in getopt |
ASTERISK-08709: Bad CALLERID after ForkCDR |
ASTERISK-08710: single step to parkcall isn't working properly although regular parking and transfers does |
ASTERISK-08711: [branch] need more 'relaxed' RFC2833 handling to interop with TI-based SIP-adapters |
ASTERISK-08712: need more 'relaxed' DTMF INFO handling to interop with TI-based SIP-adapters |
ASTERISK-08713: FIELDQTY() does not parse as expected |
ASTERISK-08714: Failure to compile on Redhat Enterprise systems: ZT_TCOP_RELEASE undeclared |
ASTERISK-08715: rtp.c: RTCP RR transmission error to, rtcp halted Success |
ASTERISK-08716: Update Makefile to show example of 'systemname' option |
ASTERISK-08717: moh-class set with SetMusicOnHold() overwrites the one set with 'm' option of Dial() |
ASTERISK-08718: [patch] Asterisk to GoogleTalk client communications fail |
ASTERISK-08719: [patch] User has to wait for ResponseTimeout before extension is dialed |
ASTERISK-08720: Patch to correctly negotiate DTMF mode when it's set to auto or SIPDtmfMode app is used |
ASTERISK-08721: can not forward Old voicemail messages |
ASTERISK-08722: Created voicemail can not be forwarded to * seperated list |
ASTERISK-08723: app_dial dosn't set DIALSTATUS if technology is missed. |
ASTERISK-08724: prepend should not mix the current voicemail file |
ASTERISK-08725: Core dumped when a Zap channel being 'Redirect'ed is hung-up |
ASTERISK-08726: chan_misdn: Redirecting no. (& reason) are not set on dialout |
ASTERISK-08727: ISDN user-to-user information field read/write |
ASTERISK-08728: ISDN user-to-user information field read/write |
ASTERISK-08729: Minor Compilation Errors on zaptel svn branch 2083 |
ASTERISK-08730: Zaptel is not compiling, something related to xbus-core.c - xbus-core.o |
ASTERISK-08731: Errors while compiling svn 53142 |
ASTERISK-08732: Directed pickup fails, No target channel |
ASTERISK-08733: GUI don't work on SVN version |
ASTERISK-08734: core dumps on 302 forward event when forwarding ignoring set |
ASTERISK-08735: [patch] AMI Status action can retunt two Responses |
ASTERISK-08736: T.38 issue. |
ASTERISK-08737: Add some flexibility for AgentCallbackLogin |
ASTERISK-08738: Transfer and Variables |
ASTERISK-08739: missing audio for sayduration parameter |
ASTERISK-08740: [patch] Configurable voicemail option prompts |
ASTERISK-08741: [patch] no callerid on incoming calls |
ASTERISK-08742: No Idle state change message when call is transferred |
ASTERISK-08743: Asterisk crashes without notice when sending a 10 digit CID to a VoIP provider (Cisco GW) |
ASTERISK-08744: insecure=very not work |
ASTERISK-08745: [patch] memory leaks with IAX realtime |
ASTERISK-08746: clear up the description of 'sendvoicemail' and 'dialout' parameters |
ASTERISK-08747: ExtensionStatus is no longer provided via the Manager interface |
ASTERISK-08748: chan_skinny randomly crashing server |
ASTERISK-08749: [patch] fix incorrect quotation of To: address of email notification |
ASTERISK-08750: stop application |
ASTERISK-08751: [patch] reloading a keyword e.g. "extconfig" prevents further reloads |
ASTERISK-08752: incoming call can be routed only to one extension |
ASTERISK-08753: [patch] Add Georgian support for say.c |
ASTERISK-08754: problem with chan_zap or chan_iax2 |
ASTERISK-08755: e-mail attachment contains CRLF within MIME (base64) data |
ASTERISK-08756: raw asterisk manager interface does not reply to querys |
ASTERISK-08757: SpeechBackground: Stop Speech Rec after DTMF received |
ASTERISK-08758: rtp.c not properly setting outbound DTMF payload type |
ASTERISK-08759: DTMF not registered from called party to 3rd call party in 3-way call |
ASTERISK-08760: Compilation warnings on kernel 2.4.27-3-386 |
ASTERISK-08761: SIP message retransmission time too short |
ASTERISK-08762: configure --with-imap fails on FC6/CentOS4/RHEL4 |
ASTERISK-08763: [patch] Queue description text clean-up |
ASTERISK-08764: Some of extensively used zaptel channels got blocked (become silent until asterisk restart) |
ASTERISK-08765: Strange audio problems (distortion, truncation...) on Intel G965 / Core2Duo system |
ASTERISK-08766: Calling Rules have stopped working |
ASTERISK-08767: configure ignores --without-oss --without-h323 |
ASTERISK-08768: Garbled sound with speex and asterisk 1.4 |
ASTERISK-08769: ./bootstrap.sh does not run |
ASTERISK-08770: Provide OPAL support for chan_h323 |
ASTERISK-08771: Call queues do not work when configured via web interface |
ASTERISK-08772: Calling Rules do not work correctly when specified via web interface |
ASTERISK-08773: chan_cellphone client |
ASTERISK-08774: configure --with-snmp uses net-snmp-config --agent-libs instead of --libs |
ASTERISK-08775: makeopts file missing ${prefix} |
ASTERISK-08776: configure --with sqlite fails on FC6/RHEL4/CentOS4 |
ASTERISK-08777: Unable to use Dynamic Features |
ASTERISK-08778: realtime optimisation |
ASTERISK-08779: Asterisk says it cannot find a required config file (contactinfo.conf) |
ASTERISK-08780: Asterisk and Zaptel have 1.4-current.tar.gz files, but not libpri |
ASTERISK-08781: IRQ clashing problem |
ASTERISK-08782: using AMD with no params crash asterisk |
ASTERISK-08783: DUNDi ignoring incoming replies when system is busy |
ASTERISK-08784: $RTPAUDIOQOS dosn't report |
ASTERISK-08785: Sipura 2000 cannot authenticate when both sip identities are in use |
ASTERISK-08786: ast_best_codec knows nothing about G722 |
ASTERISK-08787: [patch] Add confirmation of forwarded-to user via name or extension when forwarding a voicemail to another mailbox |
ASTERISK-08788: install.html fails with zapscan.conf warning |
ASTERISK-08789: [patch] Provide colors for daeminized asterisk |
ASTERISK-08790: Realtime mode in chan_sip.c add ipsvr for table sip_conf |
ASTERISK-08791: Sangoma Wanpipe Drivers unable to patch Zaptel SVN 2164 |
ASTERISK-08792: version.h contains ASTERISK_VERSION_NUM not prefixed with "0" |
ASTERISK-08793: ExtensionState remains 'Hold' after putting a call off hold |
ASTERISK-08794: [patch] Updated help text for app_record |
ASTERISK-08795: [patch] poor handling of dropped IMAP connections |
ASTERISK-08796: [patch] extend SMDI support to chan_sip |
ASTERISK-08797: ChanIsAvail returns positive ${AVAILORIGCHAN} if checking non existent IAX peer |
ASTERISK-08798: chan_sip causes spurious DNS lookups on compound hints |
ASTERISK-08799: NI-2 'Operator System Access' IE (0x01) not implemented? |
ASTERISK-08800: chanspy causes segfault |
ASTERISK-08801: After svn update sip host not work |
ASTERISK-08802: Segmentation fault in socket_process |
ASTERISK-08803: Error at call from h323 to SIP |
ASTERISK-08804: Error at call from h323 to SIP |
ASTERISK-08805: Can't be activated without mISDN-user 1.1.0 |
ASTERISK-08806: Cannot make compatible if video codecs do not match and audio codecs require transcoding |
ASTERISK-08807: transfered call does not edn until timeout when closed from the transferee |
ASTERISK-08808: [patch] Asterisk fails to pass in-band DTMF to far end during PROGRESS state. |
ASTERISK-08809: [patch] agi dumphtml has incorrect closing tag |
ASTERISK-08810: Retrieve peer address of inbound call. |
ASTERISK-08811: [patch] Search by first name does not match short names |
ASTERISK-08812: Recordings tab does not store list of recordings |
ASTERISK-08813: user pin doesnt work like expected |
ASTERISK-08814: svn last check report |
ASTERISK-08815: [patch] QSIG ROSE-12 and ROSE-13 support |
ASTERISK-08816: [patch] Playback(<file>|noanswer) and in-band info are not working with chan_skinny |
ASTERISK-08817: Going from hold to unhold status fails with Uniden phones |
ASTERISK-08818: The restart text "Restarting Asterisk NOW..." might confuse someone |
ASTERISK-08819: Locking strategy document |
ASTERISK-08820: Most heavily used Zap channels duying (become silent) after several days of operation |
ASTERISK-08821: One step parking works only in the first call |
ASTERISK-08822: [patch] print the raw read/write format |
ASTERISK-08823: Asterisk crashes randomly under heavy load with GSM->G.729 conversion |
ASTERISK-08824: Registration returned -1: Invalid argument with Wengo |
ASTERISK-08825: Bad request protocol Packet |
ASTERISK-08826: Agent in queue unavailable? |
ASTERISK-08827: [patch] app_queue description is missing membername parameter |
ASTERISK-08828: queue show and queue show <queuename> not working |
ASTERISK-08829: When caling queue, music on hold is played but not constant |
ASTERISK-08830: Install is not using the right kernel version in /lib/modules |
ASTERISK-08831: Support for autentication through LDAP server |
ASTERISK-08832: CLI bad core show command crash Asterisk |
ASTERISK-08833: Command to show zaptel verison |
ASTERISK-08834: calling rules do not appear. Loading screen constantly appears in the bar above. |
ASTERISK-08835: [patch] dnsmgr is not refreshed upon peer usage in realtime |
ASTERISK-08836: Realtime change of host in iaxpeers does not force a change in the peers without reload |
ASTERISK-08837: Calls arriving into the dialplan from chan_zap with a zero length Called Number IE fail to goto 's' priority of assigned context |
ASTERISK-08838: put on field regserver, exactly the ip where sip was registered |
ASTERISK-08839: chan_zap problem after reload |
ASTERISK-08840: Incoming calls rejected |
ASTERISK-08841: Coredump from 1.4-svn in ast_channel_free() |
ASTERISK-08842: bindaddr=0.0.0.0 no bind to eth0:1 |
ASTERISK-08843: [patch] know which file format is currently playing |
ASTERISK-08844: [patch] conf_free freed conf |
ASTERISK-08845: ChanSpy Crash |
ASTERISK-08846: MWI on Avaya phones does not work |
ASTERISK-08847: Early media in SIP |
ASTERISK-08848: Can't transfer direct to Voicemail |
ASTERISK-08849: voicemail email subjetc and voice attachment id difference |
ASTERISK-08850: [patch] Make sla.conf.sample easier to read |
ASTERISK-08851: asterisk takes a lot of time loading/reloading if not connected to internet |
ASTERISK-08852: some sound tarballs arent working |
ASTERISK-08853: Source of call blown away when setting Caller ID on PRI |
ASTERISK-08854: Log files are not being properly flushed |
ASTERISK-08855: transfer to an IAX channel fails if transferer is P2P-bridged to a transferee |
ASTERISK-08856: [patch] RTP packetization won't work under P2P bridging mode |
ASTERISK-08857: [patch] French voicemail is not really french |
ASTERISK-08858: [patch] followme for french |
ASTERISK-08859: [patch] A few problems in safe_asterisk |
ASTERISK-08860: code updates break ODBC and IMAP storage options |
ASTERISK-08861: [patch] Agent logoff soft not working |
ASTERISK-08862: properly include lock.h in utils.c |
ASTERISK-08863: [patch] small patch for ast_translate |
ASTERISK-08864: Update to dialing rule fails |
ASTERISK-08865: [patch] Asterisk duplicates results for enumlookups |
ASTERISK-08866: Voicemail attachments not working in asterisk 1.4 |
ASTERISK-08867: Compile-time failure in codec_zap.c |
ASTERISK-08868: Asterisk server crash during Passive Listening - Core segmentation fault |
ASTERISK-08869: Segementation fault. Crashes |
ASTERISK-08870: [patch] calls in queue are blocked untill it is first call in queue |
ASTERISK-08871: asterisk-gui, make checkconfig displays wrong URL |
ASTERISK-08872: Zaptel TDM400P. No Analog ports detected. Duplicate lines in zapata.conf and zaptel.conf |
ASTERISK-08873: core show file version empty |
ASTERISK-08874: nested/redundant #if !defined(LOW_MEMORY) |
ASTERISK-08875: crash svn version |
ASTERISK-08876: Make fails at codec_zap |
ASTERISK-08877: Not setting input[output]_device in alsa.conf causes a crash |
ASTERISK-08878: Zaptel trunk doesn't compile on centos 3 (kernel 2.4.21-47.0.1.ELsmp) |
ASTERISK-08879: [PATCH] Background app fails to default language if context specified |
ASTERISK-08880: Placing a call on hold sends two INVITES in specific cases (with Cisco 2,811 fateway) |
ASTERISK-08881: app_voicemail crashes when replication server falls down |
ASTERISK-08882: Function LANGUAGE not registered |
ASTERISK-08883: cannot build zaptel |
ASTERISK-08884: [patch] Clearglobalvars option does not function on dialplan reload |
ASTERISK-08885: asterisk sending text file attachments instead of wav attachment |
ASTERISK-08886: Unable to create a custom trunk by a specified name |
ASTERISK-08887: Attempting to unselect res_jabber in menuselect locks menuselect |
ASTERISK-08888: coredump on incoming IAX2 call |
ASTERISK-08889: [patch] adapt code to see all formats file |
ASTERISK-08890: chan_skinny doesn't periodically update time on phone |
ASTERISK-08891: [patch] native bridging support for chan_skinny |
ASTERISK-08892: [patch] check for frame before duping it |
ASTERISK-08893: Goto does not proceed to next prio if jump fails |
ASTERISK-08894: Asterisk crashes as soon as chan_misdn.so is unloaded |
ASTERISK-08895: Wrong FROM header |
ASTERISK-08896: Voice Mail Attachments Not Displaying As Attachments |
ASTERISK-08897: Core Dump in pbx_retrieve_variable |
ASTERISK-08898: Phone based forward will cause Asterisk to stop responding |
ASTERISK-08899: Eval leaks stack data on the end of the result string |
ASTERISK-08900: AsteriskNOW IAX users are not callable |
ASTERISK-08901: Zaptel fails when compiling V1.4 SVN branch |
ASTERISK-08902: Extend RRMEMORY strategy to use penalty (1.4.0 and 1.2.15) |
ASTERISK-08903: HDLC Bad FCS(8) on Primary D-Channel of Span 1 |
ASTERISK-08904: asterisk segfaulted in pbx.c |
ASTERISK-08905: SIP NOTIFY messages for hints are sent with wrong request-URI |
ASTERISK-08906: Segfault in res_odbc w/ pgcluster when replication server falls down |
ASTERISK-08907: Segfault in cdr_odbc |
ASTERISK-08908: sip CLI commands disappear |
ASTERISK-08909: In use status not correct for sip queue members |
ASTERISK-08910: Asterisk Crash's when callfoward from linksys 941 is activated |
ASTERISK-08911: GUI access to enable debug logging |
ASTERISK-08912: Segfault on transfers from an incoming IAX2 or Zap, towards a Queue with Agents, through a Local Dial |
ASTERISK-08913: [patch] Background DTMF escape to given context broken |
ASTERISK-08914: Call parking causes crash/deadlock |
ASTERISK-08915: asterisk loosing networking |
ASTERISK-08916: Asterisk crashes when a sip phone answers the Page |
ASTERISK-08917: Single SIP packet can reproducibly crash Asterisk |
ASTERISK-08918: [patch] Incorrect handling of zero-length frames for codecs capable of native PLC |
ASTERISK-08919: AEL parses MYSQL wrong |
ASTERISK-08920: Call-Limit Counter can be easily broken |
ASTERISK-08921: [patch] answeronpolarityswith is not working after a reload |
ASTERISK-08922: [patch] Adding language for core show channels and dumpchan |
ASTERISK-08923: IF does not work when value contains : (colon) |
ASTERISK-08924: make menuselect => core-sounds not ENglish => file not found |
ASTERISK-08925: Channel/Thread deadlock with heavy in-bound traffic on PRI - GLARE! |
ASTERISK-08926: Crash while picking up a parked call |
ASTERISK-08927: [patch] update for core* |
ASTERISK-08928: How to use CHANNEL(language) in extensions.conf ? |
ASTERISK-08929: ooGetMsgTypeText() returns incorrect text representation on message types above OOFacility |
ASTERISK-08930: Zaptel 1.2.15 not compiling on RedHat 9 |
ASTERISK-08931: marker bit lost in bridge_p2p_rtp_write |
ASTERISK-08932: null pointer dereference in res_jabber.c after "Resource (null) of buddy ... not found" |
ASTERISK-08933: GUI: hardcoded path to gui_sysinfo inside sysinfo.html |
ASTERISK-08934: Crash when loading queue via realtime. |
ASTERISK-08935: Not enough information about security issues. |
ASTERISK-08936: Crash with SLA |
ASTERISK-08937: Voicemail does not play default greeting when no options specified |
ASTERISK-08938: Crash under load from multiple SIP calls |
ASTERISK-08939: [patch] Zaptel build no longer honours HOSTCC |
ASTERISK-08940: [patch] Memory Corruption on SMP systems causes Kernel Panic |
ASTERISK-08941: race condition in sip hangup with reinvited media |
ASTERISK-08942: Can't Connect two Broadvoice accounts |
ASTERISK-08943: SVN 1.2 Rev 57962 Won't Build |
ASTERISK-08944: 'r' option disable RTP early bridge problem |
ASTERISK-08945: followme application is not respecting the channel language in order to play localized audios |
ASTERISK-08946: cdr_odbc causes segfault when odbc database becomes unavailable |
ASTERISK-08947: Wrong behaviour in Asterisk Base64 conversion routines |
ASTERISK-08948: cant unload chan_sip |
ASTERISK-08949: [patch] MYSQL Stored Procedures |
ASTERISK-08950: In voicemail.conf sendmail => date english format |
ASTERISK-08951: [patch] Make asterisk push an event to the manager in the case of a transfer. |
ASTERISK-08952: [patch] Eyebeam cannot renew subscriptions for presence info |
ASTERISK-08953: [patch] Early bridge is performed on channels with incompatible codecs when directrtpsetup=yes |
ASTERISK-08954: RTPtimeout still considered in T.38 passthrough call /ast1.4.1/ |
ASTERISK-08955: core dump on FreeBSD 6.2 with LOW MEMORY enabled when various manager commands are issued |
ASTERISK-08956: app_directory does not seem to use recorded greetings if using odbc voicemail storage |
ASTERISK-08957: sip doesnt bind to all |
ASTERISK-08958: vm-first spanish sound is wrong |
ASTERISK-08959: app_directory crashes after recent ODBC changes |
ASTERISK-08960: zaptel 1.2.15 requires kernel 2.6.17 or better for gfp_t |
ASTERISK-08961: [branch] [patch] No way to create actual macros in AEL2. |
ASTERISK-08962: Asterisk Registering to other SIP servers on non-default port. |
ASTERISK-08963: Pattern matching occurs at every sequence number of the dial plan |
ASTERISK-08964: [Patch] voicemail permissions fixes |
ASTERISK-08965: vprintk availability in kernels < 2.6.9 |
ASTERISK-08966: compilation breaks with DBUSYDETECT_MARTIN and DBUSYDETECT_TONEONLY enabled |
ASTERISK-08967: SLA continue to ring after hangup |
ASTERISK-08968: Blind transfers are parsed by [default], but attended transfers work as expected |
ASTERISK-08969: [patch] Asterisk leaves lingering UDP ports until no more ports available |
ASTERISK-08970: [patch] Calling bad SQL statement from func_odbc causes crash |
ASTERISK-08971: [branch] manager seems to not flush it's eventq like it should |
ASTERISK-08972: [patch] sip attended transfer - xfersound |
ASTERISK-08973: when stress-testing load/unloading modules, * deadlocks |
ASTERISK-08974: [patch] added support for namealias and added support for searching by first and last |
ASTERISK-08975: ZT_EVENT_REMOVED addition causes failure [patch] |
ASTERISK-08976: Can't build HPEC into 1.4 branch yet |
ASTERISK-08977: Jitterbuffer is abstracted from IAX2, but there is no 'core show netstats' |
ASTERISK-08978: [patch] Support for Cisco 7935 |
ASTERISK-08979: Channel variables are not available in 'h' extension if channel goes zombie |
ASTERISK-08980: soxmix removed from debian sid, need sox -m to do the same in Monitor application |
ASTERISK-08981: [patch] typo in help |
ASTERISK-08982: [patch] make message button work on skinny |
ASTERISK-08983: Asterisk silently loses DTMF digits when sending them through ast_dtmf_stream() |
ASTERISK-08984: [patch] User attributes in From tag |
ASTERISK-08985: Coding problem or a bug in Asterisk(AGI) |
ASTERISK-08986: pickup_exec: No target channel found for |
ASTERISK-08987: [patch] OS X/gcc inline optimization incompatability in 1.2.16 |
ASTERISK-08988: [patch] insecure && ~sipregs == Failed to authenticate |
ASTERISK-08989: [patch] fix totalAnalysisTime to handle periods of no channel activity |
ASTERISK-08990: [patch] Eliminate the comment related global vars from main/config.c |
ASTERISK-08991: Snom Call Pickup Of Subscribed Extensions |
ASTERISK-08992: make the app_dial resources configurable when returning back from a parking timeout |
ASTERISK-08993: [Patch] SMDI features |
ASTERISK-08994: Asterisk crash related to the manager and call origination |
ASTERISK-08995: Voice mail prompts in languages other than English play English digits and dates |
ASTERISK-08996: [patch] COMPLETECALLER event logs channel identifier instead of membername |
ASTERISK-08997: Misleading message for dealing with temporary greetings |
ASTERISK-08998: Can't repark calls using one touch park |
ASTERISK-08999: Greetings stored in ODBC voicemail don't play [patch] |